In coded packetized communication such as a Voice Over Internet Protocol (VoIP) system, one or some voice frame data, which are obtained by encoding a voice signal, are gathered to form a packet. After adding some further information to the packet, such as generation time, sequence number etc., the latter is transmitted along a transmission path, e.g. the Internet. The packets are successively transmitted along the transmission path and arrive at a processing point, e.g. to a receiver. Typically, the processing point is provided with a buffer (queue) for re-arranging the packets received in accordance with their transmission time, so that to overcome different arrival delays resulting from various network problems such as congestion of various links, and then the received packets are decoded.
However, suppose a packet does not arrive on time at the reception buffer. Typically, the information included in that missing packet is derived by interpolation processing using the prior frames received, a process which is called error concealment processing.
Several attempts were made to overcome these problems and enable reception of the signal in a good quality. Among the attempts made is US 20020169859A1 which describes a voice decoding apparatus with packet error resistance, which, in case a packet is not received at the appropriate time when it should be decoded, the decoded signal and a filter memory value would be calculated at that time by using a concealment processing. However, in case a packet is later received (a delayed packet), the packet would nevertheless be used in recalculating the filter memory value for frames that were formed later than the time stamp of the lost packet. This way, it would be possible to reduce/remove the long-term deterioration effect caused by the concealment processing in the filter memory value. However, no solution is provided by this publication to many cases where the packet recalculation of the filter memory value cannot be made.
WO 0230098 describes a method whereby voice is sampled and encoded to produce data that represents speech prior to its transmission. Adaptive multi-rate (AMR) speech codecs represent generation of coding algorithms that are designed to work with inaccurate transport channels, such as wireless transmission channels. The AMR speech codec has built-in mechanisms that make it tolerant to a certain level of bit errors introduced by the transport channel. Therefore, would be possible to restore the original speech with some degradation even though the coded speech is received with some bit errors.
In a publication entitled “Packet Loss and Control for Voice Transmission over the Internet” by Henning Sanneck, GMD Research Series No. 8/2000, of GMD—Forschungszentrum Informationtechnik GmbH, a further step was made. Relying on the fact that some of the coded voice frames, namely, frames where there has been a transition state form unvoiced signal to a voiced signal, comprise more important information than other frames, it has been suggested to provide these frames with extra protection. Two solutions were proposed in this publication. The first, to attach a replica of each of the so-called important frames that is carried by the N-th packets, to the N+2th packets, so that if such an Nth packet is lost, the important packet can still be regenerated from its replica carried by the N+2th packet The other solution suggested, is, that a XOR operation is carried on the Nth and N+1th packets, irrespective of the information comprised therein, and the result obtain from this operation is attached to the N+2th packet. Still, both these solutions have their drawbacks as they introduce a delay in the regenerated packet, and more importantly, they lead to a waste of bandwidth consumed by carrying the replicas of all important frames according to the first solution or even the bandwidth used for carrying the results of all those XOR operations